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[Freeswitch-users] how to add new user for external profile


 
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pankajanand18 at gmail...
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PostPosted: Wed Sep 16, 2009 9:18 am    Post subject: [Freeswitch-users] how to add new user for external profile Reply with quote

hi ,  i m very new to the FreeSwitch..
can any one tell me how to add a new user. 
i have already tried creating a new user by creating a $INSTALL_DIR/conf/directory/default/pankaj.xml :


<include>
  <user id="pankaj">
    <params>
      <param name="password" value="pankaj"/>
      <param name="vm-password" value="pankaj"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="pankaj"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension pankaj"/>
      <variable name="effective_caller_id_number" value="pankaj"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


but when i try to connect it using , the softphone shows  forbidden.
Can anyone tell me where i am making a mistake.


with regards
Pankaj anand
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tculjaga at gmail.com
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PostPosted: Wed Sep 16, 2009 9:28 am    Post subject: [Freeswitch-users] how to add new user for external profile Reply with quote

FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.


freeswitch.xml:
  <section name="directory" description="User Directory">
    <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>



Than, you need to check sip profiles. By default FS will accept registrations on internal profiles only... so you should enable it on the external as well.


look at this portion of your adequate sip profile:

   <!-- this lets anything register -->
    <!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
    <!-- <param name="accept-blind-reg" value="true"/> -->

    <!-- accept any authentication without actually checking (not a good feature for most people) -->
    <!-- <param name="accept-blind-auth" value="true"/> -->

    <!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
    <!-- <param name="suppress-cng" value="true"/> -->

    <!--TTL for nonce in sip auth-->
    <param name="nonce-ttl" value="60"/>
    <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
        that the originator is using-->
    <!--<param name="disable-transcoding" value="true"/>-->
    <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
    <!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
    <param name="auth-calls" value="$${internal_auth_calls}"/>
    <!-- Force the user and auth-user to match. -->
    <param name="inbound-reg-force-matching-username" value="true"/>
    <!-- on authed calls, authenticate *all* the packets not just invite -->
    <param name="auth-all-packets" value="false"/>
    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
    <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
    <!-- rtp inactivity timeout -->
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <!-- VAD choose one (out is a good choice); -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <!-- <param name="vad" value="both"/> -->
    <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
    <!--
        These are enabled to make the default config work better out of the box.
        If you need more than ONE domain you'll need to not use these options.

    -->
    <!--all inbound reg will look in this domain for the users -->
    <param name="force-register-domain" value="$${domain}"/>
    <!--all inbound reg will stored in the db using this domain -->
    <param name="force-register-db-domain" value="$${domain}"/>
    <!--force suscription expires to a lower value than requested-->
    <!--<param name="force-subscription-expires" value="60"/>-->
    <!-- disable register and transfer which may be undesirable in a public switch -->
    <!--<param name="disable-transfer" value="true"/>-->
    <!--<param name="disable-register" value="true"/>-->



Just make sure you use correct IP_ADDRESS:PORT to match the correct profile

vars.xml:

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>


  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>


T.


On Wed, Sep 16, 2009 at 11:29 AM, pankaj anand <pankajanand18@gmail.com (pankajanand18@gmail.com)> wrote:
Quote:
hi ,  i m very new to the FreeSwitch..
can any one tell me how to add a new user. 
i have already tried creating a new user by creating a $INSTALL_DIR/conf/directory/default/pankaj.xml :


<include>
  <user id="pankaj">
    <params>
      <param name="password" value="pankaj"/>
      <param name="vm-password" value="pankaj"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="pankaj"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension pankaj"/>
      <variable name="effective_caller_id_number" value="pankaj"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


but when i try to connect it using , the softphone shows  forbidden.
Can anyone tell me where i am making a mistake.


with regards
Pankaj anand








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